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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
| commit | 2aa4a82499d4becd2284cdb482213d541b8804dd (patch) | |
| tree | b80bf8bf13c3766139fbacc530efd0dd9d54394c /dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html | |
| parent | Initial commit. (diff) | |
| download | firefox-upstream.tar.xz firefox-upstream.zip | |
Adding upstream version 86.0.1.upstream/86.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html')
| -rw-r--r-- | dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html | 86 |
1 files changed, 86 insertions, 0 deletions
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html new file mode 100644 index 0000000000..a68f148322 --- /dev/null +++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html @@ -0,0 +1,86 @@ +<!DOCTYPE HTML> +<html> +<head> + <script type="application/javascript" src="pc.js"></script> +</head> +<body> +<pre id="test"> +<script type="application/javascript"> + createHTML({ + bug: "1395853", + title: "Verify audio content over WebRTC for every audio codec", + }); + + // We match the format member against the sdp to figure out the payload type, + // So all other present codecs can be removed. + const codecs = [ "opus", "G722", "PCMU", "PCMA" ]; + + async function testAudioCodec(options = {}, codec) { + // sdputils checks for opus as part of its sdp sanity test + options.opus = codec == "opus"; + + let test = new PeerConnectionTest(options); + test.setMediaConstraints([{audio: true}], []); + + test.chain.insertBefore("PC_LOCAL_SET_LOCAL_DESCRIPTION", [ + function PC_LOCAL_FILTER_OUT_CODECS() { + let otherCodec = codecs.find(c => c != codec); + let otherId = sdputils.findCodecId(test.originalOffer.sdp, otherCodec); + + let id = sdputils.findCodecId(test.originalOffer.sdp, codec); + test.originalOffer.sdp = + sdputils.removeAllButPayloadType(test.originalOffer.sdp, id); + + ok(!test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${otherId}[^0-9]`, "gi")), + `Other codec ${otherId} should be removed after filtering`); + ok(test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${id}[^0-9]`, "gi")), + `Tested codec ${id} should remain after filtering`); + + for (let c of codecs.filter(c => c != codec)) { + // Remove rtpmaps for the other codecs so sdp sanity tests pass. + let id = sdputils.findCodecId(test.originalOffer.sdp, c); + test.originalOffer.sdp = + sdputils.removeRtpMapForPayloadType(test.originalOffer.sdp, id); + } + + ok(!test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${otherId}.*\\r\\n`, "gi")), + `Rtpmap of other codec ${otherId} should be removed after filtering`); + ok(test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${id}.*\\r\\n`, "gi")), + `Rtpmap of tested codec should remain after filtering`); + }, + ]); + + test.chain.append([ + async function CHECK_AUDIO_FLOW() { + try { + await test.pcRemote.checkReceivingToneFrom(new AudioContext(), test.pcLocal); + ok(true, "input and output audio data matches"); + } catch(e) { + ok(false, `No audio flow: ${e}`); + } + }, + ]); + + // This inlines test.run(), to allow for multiple tests to run. + test.updateChainSteps(); + await test.chain.execute(); + await test.close(); + } + + runNetworkTest(async (options) => { + for (let codec of codecs) { + info(`Testing audio for codec ${codec}`); + try { + await testAudioCodec(options, codec); + } catch(e) { + ok(false, `Error in test for codec ${codec}: ${e}\n${e.stack}`); + } + info(`Tested audio for codec ${codec}`); + } + + networkTestFinished(); + }); +</script> +</pre> +</body> +</html> |